/* audioopmodule - Module to detect peak values in arrays */ #define PY_SSIZE_T_CLEAN #include "Python.h" #if defined(__CHAR_UNSIGNED__) #if defined(signed) /* This module currently does not work on systems where only unsigned characters are available. Take it out of Setup. Sorry. */ #endif #endif static const int maxvals[] = {0, 0x7F, 0x7FFF, 0x7FFFFF, 0x7FFFFFFF}; /* -1 trick is needed on Windows to support -0x80000000 without a warning */ static const int minvals[] = {0, -0x80, -0x8000, -0x800000, -0x7FFFFFFF-1}; static const unsigned int masks[] = {0, 0xFF, 0xFFFF, 0xFFFFFF, 0xFFFFFFFF}; static int fbound(double val, double minval, double maxval) { if (val > maxval) { val = maxval; } else if (val < minval + 1.0) { val = minval; } /* Round towards minus infinity (-inf) */ val = floor(val); /* Cast double to integer: round towards zero */ return (int)val; } /* Code shamelessly stolen from sox, 12.17.7, g711.c ** (c) Craig Reese, Joe Campbell and Jeff Poskanzer 1989 */ /* From g711.c: * * December 30, 1994: * Functions linear2alaw, linear2ulaw have been updated to correctly * convert unquantized 16 bit values. * Tables for direct u- to A-law and A- to u-law conversions have been * corrected. * Borge Lindberg, Center for PersonKommunikation, Aalborg University. * bli@cpk.auc.dk * */ #define BIAS 0x84 /* define the add-in bias for 16 bit samples */ #define CLIP 32635 #define SIGN_BIT (0x80) /* Sign bit for an A-law byte. */ #define QUANT_MASK (0xf) /* Quantization field mask. */ #define SEG_SHIFT (4) /* Left shift for segment number. */ #define SEG_MASK (0x70) /* Segment field mask. */ static const int16_t seg_aend[8] = { 0x1F, 0x3F, 0x7F, 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF }; static const int16_t seg_uend[8] = { 0x3F, 0x7F, 0xFF, 0x1FF, 0x3FF, 0x7FF, 0xFFF, 0x1FFF }; static int16_t search(int16_t val, const int16_t *table, int size) { int i; for (i = 0; i < size; i++) { if (val <= *table++) return (i); } return (size); } #define st_ulaw2linear16(uc) (_st_ulaw2linear16[uc]) #define st_alaw2linear16(uc) (_st_alaw2linear16[uc]) static const int16_t _st_ulaw2linear16[256] = { -32124, -31100, -30076, -29052, -28028, -27004, -25980, -24956, -23932, -22908, -21884, -20860, -19836, -18812, -17788, -16764, -15996, -15484, -14972, -14460, -13948, -13436, -12924, -12412, -11900, -11388, -10876, -10364, -9852, -9340, -8828, -8316, -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, -876, -844, -812, -780, -748, -716, -684, -652, -620, -588, -556, -524, -492, -460, -428, -396, -372, -356, -340, -324, -308, -292, -276, -260, -244, -228, -212, -196, -180, -164, -148, -132, -120, -112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8, 0, 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, 876, 844, 812, 780, 748, 716, 684, 652, 620, 588, 556, 524, 492, 460, 428, 396, 372, 356, 340, 324, 308, 292, 276, 260, 244, 228, 212, 196, 180, 164, 148, 132, 120, 112, 104, 96, 88, 80, 72, 64, 56, 48, 40, 32, 24, 16, 8, 0 }; /* * linear2ulaw() accepts a 14-bit signed integer and encodes it as u-law data * stored in an unsigned char. This function should only be called with * the data shifted such that it only contains information in the lower * 14-bits. * * In order to simplify the encoding process, the original linear magnitude * is biased by adding 33 which shifts the encoding range from (0 - 8158) to * (33 - 8191). The result can be seen in the following encoding table: * * Biased Linear Input Code Compressed Code * ------------------------ --------------- * 00000001wxyza 000wxyz * 0000001wxyzab 001wxyz * 000001wxyzabc 010wxyz * 00001wxyzabcd 011wxyz * 0001wxyzabcde 100wxyz * 001wxyzabcdef 101wxyz * 01wxyzabcdefg 110wxyz * 1wxyzabcdefgh 111wxyz * * Each biased linear code has a leading 1 which identifies the segment * number. The value of the segment number is equal to 7 minus the number * of leading 0's. The quantization interval is directly available as the * four bits wxyz. * The trailing bits (a - h) are ignored. * * Ordinarily the complement of the resulting code word is used for * transmission, and so the code word is complemented before it is returned. * * For further information see John C. Bellamy's Digital Telephony, 1982, * John Wiley & Sons, pps 98-111 and 472-476. */ static unsigned char st_14linear2ulaw(int16_t pcm_val) /* 2's complement (14-bit range) */ { int16_t mask; int16_t seg; unsigned char uval; /* u-law inverts all bits */ /* Get the sign and the magnitude of the value. */ if (pcm_val < 0) { pcm_val = -pcm_val; mask = 0x7F; } else { mask = 0xFF; } if ( pcm_val > CLIP ) pcm_val = CLIP; /* clip the magnitude */ pcm_val += (BIAS >> 2); /* Convert the scaled magnitude to segment number. */ seg = search(pcm_val, seg_uend, 8); /* * Combine the sign, segment, quantization bits; * and complement the code word. */ if (seg >= 8) /* out of range, return maximum value. */ return (unsigned char) (0x7F ^ mask); else { uval = (unsigned char) (seg << 4) | ((pcm_val >> (seg + 1)) & 0xF); return (uval ^ mask); } } static const int16_t _st_alaw2linear16[256] = { -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, -2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368, -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, -22016, -20992, -24064, -23040, -17920, -16896, -19968, -18944, -30208, -29184, -32256, -31232, -26112, -25088, -28160, -27136, -11008, -10496, -12032, -11520, -8960, -8448, -9984, -9472, -15104, -14592, -16128, -15616, -13056, -12544, -14080, -13568, -344, -328, -376, -360, -280, -264, -312, -296, -472, -456, -504, -488, -408, -392, -440, -424, -88, -72, -120, -104, -24, -8, -56, -40, -216, -200, -248, -232, -152, -136, -184, -168, -1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184, -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, -688, -656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848, 5504, 5248, 6016, 5760, 4480, 4224, 4992, 4736, 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784, 2752, 2624, 3008, 2880, 2240, 2112, 2496, 2368, 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392, 22016, 20992, 24064, 23040, 17920, 16896, 19968, 18944, 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136, 11008, 10496, 12032, 11520, 8960, 8448, 9984, 9472, 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568, 344, 328, 376, 360, 280, 264, 312, 296, 472, 456, 504, 488, 408, 392, 440, 424, 88, 72, 120, 104, 24, 8, 56, 40, 216, 200, 248, 232, 152, 136, 184, 168, 1376, 1312, 1504, 1440, 1120, 1056, 1248, 1184, 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696, 688, 656, 752, 720, 560, 528, 624, 592, 944, 912, 1008, 976, 816, 784, 880, 848 }; /* * linear2alaw() accepts a 13-bit signed integer and encodes it as A-law data * stored in an unsigned char. This function should only be called with * the data shifted such that it only contains information in the lower * 13-bits. * * Linear Input Code Compressed Code * ------------------------ --------------- * 0000000wxyza 000wxyz * 0000001wxyza 001wxyz * 000001wxyzab 010wxyz * 00001wxyzabc 011wxyz * 0001wxyzabcd 100wxyz * 001wxyzabcde 101wxyz * 01wxyzabcdef 110wxyz * 1wxyzabcdefg 111wxyz * * For further information see John C. Bellamy's Digital Telephony, 1982, * John Wiley & Sons, pps 98-111 and 472-476. */ static unsigned char st_linear2alaw(int16_t pcm_val) /* 2's complement (13-bit range) */ { int16_t mask; int16_t seg; unsigned char aval; /* A-law using even bit inversion */ if (pcm_val >= 0) { mask = 0xD5; /* sign (7th) bit = 1 */ } else { mask = 0x55; /* sign bit = 0 */ pcm_val = -pcm_val - 1; } /* Convert the scaled magnitude to segment number. */ seg = search(pcm_val, seg_aend, 8); /* Combine the sign, segment, and quantization bits. */ if (seg >= 8) /* out of range, return maximum value. */ return (unsigned char) (0x7F ^ mask); else { aval = (unsigned char) seg << SEG_SHIFT; if (seg < 2) aval |= (pcm_val >> 1) & QUANT_MASK; else aval |= (pcm_val >> seg) & QUANT_MASK; return (aval ^ mask); } } /* End of code taken from sox */ /* Intel ADPCM step variation table */ static const int indexTable[16] = { -1, -1, -1, -1, 2, 4, 6, 8, -1, -1, -1, -1, 2, 4, 6, 8, }; static const int stepsizeTable[89] = { 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 }; #define GETINTX(T, cp, i) (*(T *)((unsigned char *)(cp) + (i))) #define SETINTX(T, cp, i, val) do { \ *(T *)((unsigned char *)(cp) + (i)) = (T)(val); \ } while (0) #define GETINT8(cp, i) GETINTX(signed char, (cp), (i)) #define GETINT16(cp, i) GETINTX(int16_t, (cp), (i)) #define GETINT32(cp, i) GETINTX(int32_t, (cp), (i)) #if WORDS_BIGENDIAN #define GETINT24(cp, i) ( \ ((unsigned char *)(cp) + (i))[2] + \ (((unsigned char *)(cp) + (i))[1] << 8) + \ (((signed char *)(cp) + (i))[0] << 16) ) #else #define GETINT24(cp, i) ( \ ((unsigned char *)(cp) + (i))[0] + \ (((unsigned char *)(cp) + (i))[1] << 8) + \ (((signed char *)(cp) + (i))[2] << 16) ) #endif #define SETINT8(cp, i, val) SETINTX(signed char, (cp), (i), (val)) #define SETINT16(cp, i, val) SETINTX(int16_t, (cp), (i), (val)) #define SETINT32(cp, i, val) SETINTX(int32_t, (cp), (i), (val)) #if WORDS_BIGENDIAN #define SETINT24(cp, i, val) do { \ ((unsigned char *)(cp) + (i))[2] = (int)(val); \ ((unsigned char *)(cp) + (i))[1] = (int)(val) >> 8; \ ((signed char *)(cp) + (i))[0] = (int)(val) >> 16; \ } while (0) #else #define SETINT24(cp, i, val) do { \ ((unsigned char *)(cp) + (i))[0] = (int)(val); \ ((unsigned char *)(cp) + (i))[1] = (int)(val) >> 8; \ ((signed char *)(cp) + (i))[2] = (int)(val) >> 16; \ } while (0) #endif #define GETRAWSAMPLE(size, cp, i) ( \ (size == 1) ? (int)GETINT8((cp), (i)) : \ (size == 2) ? (int)GETINT16((cp), (i)) : \ (size == 3) ? (int)GETINT24((cp), (i)) : \ (int)GETINT32((cp), (i))) #define SETRAWSAMPLE(size, cp, i, val) do { \ if (size == 1) \ SETINT8((cp), (i), (val)); \ else if (size == 2) \ SETINT16((cp), (i), (val)); \ else if (size == 3) \ SETINT24((cp), (i), (val)); \ else \ SETINT32((cp), (i), (val)); \ } while(0) #define GETSAMPLE32(size, cp, i) ( \ (size == 1) ? (int)GETINT8((cp), (i)) << 24 : \ (size == 2) ? (int)GETINT16((cp), (i)) << 16 : \ (size == 3) ? (int)GETINT24((cp), (i)) << 8 : \ (int)GETINT32((cp), (i))) #define SETSAMPLE32(size, cp, i, val) do { \ if (size == 1) \ SETINT8((cp), (i), (val) >> 24); \ else if (size == 2) \ SETINT16((cp), (i), (val) >> 16); \ else if (size == 3) \ SETINT24((cp), (i), (val) >> 8); \ else \ SETINT32((cp), (i), (val)); \ } while(0) static PyObject *AudioopError; static int audioop_check_size(int size) { if (size < 1 || size > 4) { PyErr_SetString(AudioopError, "Size should be 1, 2, 3 or 4"); return 0; } else return 1; } static int audioop_check_parameters(Py_ssize_t len, int size) { if (!audioop_check_size(size)) return 0; if (len % size != 0) { PyErr_SetString(AudioopError, "not a whole number of frames"); return 0; } return 1; } /*[clinic input] module audioop [clinic start generated code]*/ /*[clinic end generated code: output=da39a3ee5e6b4b0d input=8fa8f6611be3591a]*/ /*[clinic input] audioop.getsample fragment: Py_buffer width: int index: Py_ssize_t / Return the value of sample index from the fragment. [clinic start generated code]*/ static PyObject * audioop_getsample_impl(PyObject *module, Py_buffer *fragment, int width, Py_ssize_t index) /*[clinic end generated code: output=8fe1b1775134f39a input=88edbe2871393549]*/ { int val; if (!audioop_check_parameters(fragment->len, width)) return NULL; if (index < 0 || index >= fragment->len/width) { PyErr_SetString(AudioopError, "Index out of range"); return NULL; } val = GETRAWSAMPLE(width, fragment->buf, index*width); return PyLong_FromLong(val); } /*[clinic input] audioop.max fragment: Py_buffer width: int / Return the maximum of the absolute value of all samples in a fragment. [clinic start generated code]*/ static PyObject * audioop_max_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=e6c5952714f1c3f0 input=32bea5ea0ac8c223]*/ { Py_ssize_t i; unsigned int absval, max = 0; if (!audioop_check_parameters(fragment->len, width)) return NULL; for (i = 0; i < fragment->len; i += width) { int val = GETRAWSAMPLE(width, fragment->buf, i); /* Cast to unsigned before negating. Unsigned overflow is well- defined, but signed overflow is not. */ if (val < 0) absval = (unsigned int)-(int64_t)val; else absval = val; if (absval > max) max = absval; } return PyLong_FromUnsignedLong(max); } /*[clinic input] audioop.minmax fragment: Py_buffer width: int / Return the minimum and maximum values of all samples in the sound fragment. [clinic start generated code]*/ static PyObject * audioop_minmax_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=473fda66b15c836e input=89848e9b927a0696]*/ { Py_ssize_t i; /* -1 trick below is needed on Windows to support -0x80000000 without a warning */ int min = 0x7fffffff, max = -0x7FFFFFFF-1; if (!audioop_check_parameters(fragment->len, width)) return NULL; for (i = 0; i < fragment->len; i += width) { int val = GETRAWSAMPLE(width, fragment->buf, i); if (val > max) max = val; if (val < min) min = val; } return Py_BuildValue("(ii)", min, max); } /*[clinic input] audioop.avg fragment: Py_buffer width: int / Return the average over all samples in the fragment. [clinic start generated code]*/ static PyObject * audioop_avg_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=4410a4c12c3586e6 input=1114493c7611334d]*/ { Py_ssize_t i; int avg; double sum = 0.0; if (!audioop_check_parameters(fragment->len, width)) return NULL; for (i = 0; i < fragment->len; i += width) sum += GETRAWSAMPLE(width, fragment->buf, i); if (fragment->len == 0) avg = 0; else avg = (int)floor(sum / (double)(fragment->len/width)); return PyLong_FromLong(avg); } /*[clinic input] audioop.rms fragment: Py_buffer width: int / Return the root-mean-square of the fragment, i.e. sqrt(sum(S_i^2)/n). [clinic start generated code]*/ static PyObject * audioop_rms_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=1e7871c826445698 input=4cc57c6c94219d78]*/ { Py_ssize_t i; unsigned int res; double sum_squares = 0.0; if (!audioop_check_parameters(fragment->len, width)) return NULL; for (i = 0; i < fragment->len; i += width) { double val = GETRAWSAMPLE(width, fragment->buf, i); sum_squares += val*val; } if (fragment->len == 0) res = 0; else res = (unsigned int)sqrt(sum_squares / (double)(fragment->len/width)); return PyLong_FromUnsignedLong(res); } static double _sum2(const int16_t *a, const int16_t *b, Py_ssize_t len) { Py_ssize_t i; double sum = 0.0; for( i=0; i n, and let all sums be over i from 0 to n-1. ** ** Now, for each j in {0..N-n} we compute a factor fj so that -fj*R matches A ** as good as possible, i.e. sum( (A[j+i]+fj*R[i])^2 ) is minimal. This ** equation gives fj = sum( A[j+i]R[i] ) / sum(R[i]^2). ** ** Next, we compute the relative distance between the original signal and ** the modified signal and minimize that over j: ** vj = sum( (A[j+i]-fj*R[i])^2 ) / sum( A[j+i]^2 ) => ** vj = ( sum(A[j+i]^2)*sum(R[i]^2) - sum(A[j+i]R[i])^2 ) / sum( A[j+i]^2 ) ** ** In the code variables correspond as follows: ** cp1 A ** cp2 R ** len1 N ** len2 n ** aj_m1 A[j-1] ** aj_lm1 A[j+n-1] ** sum_ri_2 sum(R[i]^2) ** sum_aij_2 sum(A[i+j]^2) ** sum_aij_ri sum(A[i+j]R[i]) ** ** sum_ri is calculated once, sum_aij_2 is updated each step and sum_aij_ri ** is completely recalculated each step. */ /*[clinic input] audioop.findfit fragment: Py_buffer reference: Py_buffer / Try to match reference as well as possible to a portion of fragment. [clinic start generated code]*/ static PyObject * audioop_findfit_impl(PyObject *module, Py_buffer *fragment, Py_buffer *reference) /*[clinic end generated code: output=5752306d83cbbada input=62c305605e183c9a]*/ { const int16_t *cp1, *cp2; Py_ssize_t len1, len2; Py_ssize_t j, best_j; double aj_m1, aj_lm1; double sum_ri_2, sum_aij_2, sum_aij_ri, result, best_result, factor; if (fragment->len & 1 || reference->len & 1) { PyErr_SetString(AudioopError, "Strings should be even-sized"); return NULL; } cp1 = (const int16_t *)fragment->buf; len1 = fragment->len >> 1; cp2 = (const int16_t *)reference->buf; len2 = reference->len >> 1; if (len1 < len2) { PyErr_SetString(AudioopError, "First sample should be longer"); return NULL; } sum_ri_2 = _sum2(cp2, cp2, len2); sum_aij_2 = _sum2(cp1, cp1, len2); sum_aij_ri = _sum2(cp1, cp2, len2); result = (sum_ri_2*sum_aij_2 - sum_aij_ri*sum_aij_ri) / sum_aij_2; best_result = result; best_j = 0; for ( j=1; j<=len1-len2; j++) { aj_m1 = (double)cp1[j-1]; aj_lm1 = (double)cp1[j+len2-1]; sum_aij_2 = sum_aij_2 + aj_lm1*aj_lm1 - aj_m1*aj_m1; sum_aij_ri = _sum2(cp1+j, cp2, len2); result = (sum_ri_2*sum_aij_2 - sum_aij_ri*sum_aij_ri) / sum_aij_2; if ( result < best_result ) { best_result = result; best_j = j; } } factor = _sum2(cp1+best_j, cp2, len2) / sum_ri_2; return Py_BuildValue("(nf)", best_j, factor); } /* ** findfactor finds a factor f so that the energy in A-fB is minimal. ** See the comment for findfit for details. */ /*[clinic input] audioop.findfactor fragment: Py_buffer reference: Py_buffer / Return a factor F such that rms(add(fragment, mul(reference, -F))) is minimal. [clinic start generated code]*/ static PyObject * audioop_findfactor_impl(PyObject *module, Py_buffer *fragment, Py_buffer *reference) /*[clinic end generated code: output=14ea95652c1afcf8 input=816680301d012b21]*/ { const int16_t *cp1, *cp2; Py_ssize_t len; double sum_ri_2, sum_aij_ri, result; if (fragment->len & 1 || reference->len & 1) { PyErr_SetString(AudioopError, "Strings should be even-sized"); return NULL; } if (fragment->len != reference->len) { PyErr_SetString(AudioopError, "Samples should be same size"); return NULL; } cp1 = (const int16_t *)fragment->buf; cp2 = (const int16_t *)reference->buf; len = fragment->len >> 1; sum_ri_2 = _sum2(cp2, cp2, len); sum_aij_ri = _sum2(cp1, cp2, len); result = sum_aij_ri / sum_ri_2; return PyFloat_FromDouble(result); } /* ** findmax returns the index of the n-sized segment of the input sample ** that contains the most energy. */ /*[clinic input] audioop.findmax fragment: Py_buffer length: Py_ssize_t / Search fragment for a slice of specified number of samples with maximum energy. [clinic start generated code]*/ static PyObject * audioop_findmax_impl(PyObject *module, Py_buffer *fragment, Py_ssize_t length) /*[clinic end generated code: output=f008128233523040 input=2f304801ed42383c]*/ { const int16_t *cp1; Py_ssize_t len1; Py_ssize_t j, best_j; double aj_m1, aj_lm1; double result, best_result; if (fragment->len & 1) { PyErr_SetString(AudioopError, "Strings should be even-sized"); return NULL; } cp1 = (const int16_t *)fragment->buf; len1 = fragment->len >> 1; if (length < 0 || len1 < length) { PyErr_SetString(AudioopError, "Input sample should be longer"); return NULL; } result = _sum2(cp1, cp1, length); best_result = result; best_j = 0; for ( j=1; j<=len1-length; j++) { aj_m1 = (double)cp1[j-1]; aj_lm1 = (double)cp1[j+length-1]; result = result + aj_lm1*aj_lm1 - aj_m1*aj_m1; if ( result > best_result ) { best_result = result; best_j = j; } } return PyLong_FromSsize_t(best_j); } /*[clinic input] audioop.avgpp fragment: Py_buffer width: int / Return the average peak-peak value over all samples in the fragment. [clinic start generated code]*/ static PyObject * audioop_avgpp_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=269596b0d5ae0b2b input=0b3cceeae420a7d9]*/ { Py_ssize_t i; int prevval, prevextremevalid = 0, prevextreme = 0; double sum = 0.0; unsigned int avg; int diff, prevdiff, nextreme = 0; if (!audioop_check_parameters(fragment->len, width)) return NULL; if (fragment->len <= width) return PyLong_FromLong(0); prevval = GETRAWSAMPLE(width, fragment->buf, 0); prevdiff = 17; /* Anything != 0, 1 */ for (i = width; i < fragment->len; i += width) { int val = GETRAWSAMPLE(width, fragment->buf, i); if (val != prevval) { diff = val < prevval; if (prevdiff == !diff) { /* Derivative changed sign. Compute difference to last ** extreme value and remember. */ if (prevextremevalid) { if (prevval < prevextreme) sum += (double)((unsigned int)prevextreme - (unsigned int)prevval); else sum += (double)((unsigned int)prevval - (unsigned int)prevextreme); nextreme++; } prevextremevalid = 1; prevextreme = prevval; } prevval = val; prevdiff = diff; } } if ( nextreme == 0 ) avg = 0; else avg = (unsigned int)(sum / (double)nextreme); return PyLong_FromUnsignedLong(avg); } /*[clinic input] audioop.maxpp fragment: Py_buffer width: int / Return the maximum peak-peak value in the sound fragment. [clinic start generated code]*/ static PyObject * audioop_maxpp_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=5b918ed5dbbdb978 input=671a13e1518f80a1]*/ { Py_ssize_t i; int prevval, prevextremevalid = 0, prevextreme = 0; unsigned int max = 0, extremediff; int diff, prevdiff; if (!audioop_check_parameters(fragment->len, width)) return NULL; if (fragment->len <= width) return PyLong_FromLong(0); prevval = GETRAWSAMPLE(width, fragment->buf, 0); prevdiff = 17; /* Anything != 0, 1 */ for (i = width; i < fragment->len; i += width) { int val = GETRAWSAMPLE(width, fragment->buf, i); if (val != prevval) { diff = val < prevval; if (prevdiff == !diff) { /* Derivative changed sign. Compute difference to ** last extreme value and remember. */ if (prevextremevalid) { if (prevval < prevextreme) extremediff = (unsigned int)prevextreme - (unsigned int)prevval; else extremediff = (unsigned int)prevval - (unsigned int)prevextreme; if ( extremediff > max ) max = extremediff; } prevextremevalid = 1; prevextreme = prevval; } prevval = val; prevdiff = diff; } } return PyLong_FromUnsignedLong(max); } /*[clinic input] audioop.cross fragment: Py_buffer width: int / Return the number of zero crossings in the fragment passed as an argument. [clinic start generated code]*/ static PyObject * audioop_cross_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=5938dcdd74a1f431 input=b1b3f15b83f6b41a]*/ { Py_ssize_t i; int prevval; Py_ssize_t ncross; if (!audioop_check_parameters(fragment->len, width)) return NULL; ncross = -1; prevval = 17; /* Anything <> 0,1 */ for (i = 0; i < fragment->len; i += width) { int val = GETRAWSAMPLE(width, fragment->buf, i) < 0; if (val != prevval) ncross++; prevval = val; } return PyLong_FromSsize_t(ncross); } /*[clinic input] audioop.mul fragment: Py_buffer width: int factor: double / Return a fragment that has all samples in the original fragment multiplied by the floating-point value factor. [clinic start generated code]*/ static PyObject * audioop_mul_impl(PyObject *module, Py_buffer *fragment, int width, double factor) /*[clinic end generated code: output=6cd48fe796da0ea4 input=c726667baa157d3c]*/ { signed char *ncp; Py_ssize_t i; double maxval, minval; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; maxval = (double) maxvals[width]; minval = (double) minvals[width]; rv = PyBytes_FromStringAndSize(NULL, fragment->len); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); for (i = 0; i < fragment->len; i += width) { double val = GETRAWSAMPLE(width, fragment->buf, i); int ival = fbound(val * factor, minval, maxval); SETRAWSAMPLE(width, ncp, i, ival); } return rv; } /*[clinic input] audioop.tomono fragment: Py_buffer width: int lfactor: double rfactor: double / Convert a stereo fragment to a mono fragment. [clinic start generated code]*/ static PyObject * audioop_tomono_impl(PyObject *module, Py_buffer *fragment, int width, double lfactor, double rfactor) /*[clinic end generated code: output=235c8277216d4e4e input=c4ec949b3f4dddfa]*/ { signed char *cp, *ncp; Py_ssize_t len, i; double maxval, minval; PyObject *rv; cp = fragment->buf; len = fragment->len; if (!audioop_check_parameters(len, width)) return NULL; if (((len / width) & 1) != 0) { PyErr_SetString(AudioopError, "not a whole number of frames"); return NULL; } maxval = (double) maxvals[width]; minval = (double) minvals[width]; rv = PyBytes_FromStringAndSize(NULL, len/2); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); for (i = 0; i < len; i += width*2) { double val1 = GETRAWSAMPLE(width, cp, i); double val2 = GETRAWSAMPLE(width, cp, i + width); double val = val1 * lfactor + val2 * rfactor; int ival = fbound(val, minval, maxval); SETRAWSAMPLE(width, ncp, i/2, ival); } return rv; } /*[clinic input] audioop.tostereo fragment: Py_buffer width: int lfactor: double rfactor: double / Generate a stereo fragment from a mono fragment. [clinic start generated code]*/ static PyObject * audioop_tostereo_impl(PyObject *module, Py_buffer *fragment, int width, double lfactor, double rfactor) /*[clinic end generated code: output=046f13defa5f1595 input=27b6395ebfdff37a]*/ { signed char *ncp; Py_ssize_t i; double maxval, minval; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; maxval = (double) maxvals[width]; minval = (double) minvals[width]; if (fragment->len > PY_SSIZE_T_MAX/2) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); return NULL; } rv = PyBytes_FromStringAndSize(NULL, fragment->len*2); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); for (i = 0; i < fragment->len; i += width) { double val = GETRAWSAMPLE(width, fragment->buf, i); int val1 = fbound(val * lfactor, minval, maxval); int val2 = fbound(val * rfactor, minval, maxval); SETRAWSAMPLE(width, ncp, i*2, val1); SETRAWSAMPLE(width, ncp, i*2 + width, val2); } return rv; } /*[clinic input] audioop.add fragment1: Py_buffer fragment2: Py_buffer width: int / Return a fragment which is the addition of the two samples passed as parameters. [clinic start generated code]*/ static PyObject * audioop_add_impl(PyObject *module, Py_buffer *fragment1, Py_buffer *fragment2, int width) /*[clinic end generated code: output=60140af4d1aab6f2 input=4a8d4bae4c1605c7]*/ { signed char *ncp; Py_ssize_t i; int minval, maxval, newval; PyObject *rv; if (!audioop_check_parameters(fragment1->len, width)) return NULL; if (fragment1->len != fragment2->len) { PyErr_SetString(AudioopError, "Lengths should be the same"); return NULL; } maxval = maxvals[width]; minval = minvals[width]; rv = PyBytes_FromStringAndSize(NULL, fragment1->len); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); for (i = 0; i < fragment1->len; i += width) { int val1 = GETRAWSAMPLE(width, fragment1->buf, i); int val2 = GETRAWSAMPLE(width, fragment2->buf, i); if (width < 4) { newval = val1 + val2; /* truncate in case of overflow */ if (newval > maxval) newval = maxval; else if (newval < minval) newval = minval; } else { double fval = (double)val1 + (double)val2; /* truncate in case of overflow */ newval = fbound(fval, minval, maxval); } SETRAWSAMPLE(width, ncp, i, newval); } return rv; } /*[clinic input] audioop.bias fragment: Py_buffer width: int bias: int / Return a fragment that is the original fragment with a bias added to each sample. [clinic start generated code]*/ static PyObject * audioop_bias_impl(PyObject *module, Py_buffer *fragment, int width, int bias) /*[clinic end generated code: output=6e0aa8f68f045093 input=2b5cce5c3bb4838c]*/ { signed char *ncp; Py_ssize_t i; unsigned int val = 0, mask; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; rv = PyBytes_FromStringAndSize(NULL, fragment->len); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); mask = masks[width]; for (i = 0; i < fragment->len; i += width) { if (width == 1) val = GETINTX(unsigned char, fragment->buf, i); else if (width == 2) val = GETINTX(uint16_t, fragment->buf, i); else if (width == 3) val = ((unsigned int)GETINT24(fragment->buf, i)) & 0xffffffu; else { assert(width == 4); val = GETINTX(uint32_t, fragment->buf, i); } val += (unsigned int)bias; /* wrap around in case of overflow */ val &= mask; if (width == 1) SETINTX(unsigned char, ncp, i, val); else if (width == 2) SETINTX(uint16_t, ncp, i, val); else if (width == 3) SETINT24(ncp, i, (int)val); else { assert(width == 4); SETINTX(uint32_t, ncp, i, val); } } return rv; } /*[clinic input] audioop.reverse fragment: Py_buffer width: int / Reverse the samples in a fragment and returns the modified fragment. [clinic start generated code]*/ static PyObject * audioop_reverse_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=b44135698418da14 input=668f890cf9f9d225]*/ { unsigned char *ncp; Py_ssize_t i; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; rv = PyBytes_FromStringAndSize(NULL, fragment->len); if (rv == NULL) return NULL; ncp = (unsigned char *)PyBytes_AsString(rv); for (i = 0; i < fragment->len; i += width) { int val = GETRAWSAMPLE(width, fragment->buf, i); SETRAWSAMPLE(width, ncp, fragment->len - i - width, val); } return rv; } /*[clinic input] audioop.byteswap fragment: Py_buffer width: int / Convert big-endian samples to little-endian and vice versa. [clinic start generated code]*/ static PyObject * audioop_byteswap_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=50838a9e4b87cd4d input=fae7611ceffa5c82]*/ { unsigned char *ncp; Py_ssize_t i; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; rv = PyBytes_FromStringAndSize(NULL, fragment->len); if (rv == NULL) return NULL; ncp = (unsigned char *)PyBytes_AsString(rv); for (i = 0; i < fragment->len; i += width) { int j; for (j = 0; j < width; j++) ncp[i + width - 1 - j] = ((unsigned char *)fragment->buf)[i + j]; } return rv; } /*[clinic input] audioop.lin2lin fragment: Py_buffer width: int newwidth: int / Convert samples between 1-, 2-, 3- and 4-byte formats. [clinic start generated code]*/ static PyObject * audioop_lin2lin_impl(PyObject *module, Py_buffer *fragment, int width, int newwidth) /*[clinic end generated code: output=17b14109248f1d99 input=5ce08c8aa2f24d96]*/ { unsigned char *ncp; Py_ssize_t i, j; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; if (!audioop_check_size(newwidth)) return NULL; if (fragment->len/width > PY_SSIZE_T_MAX/newwidth) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); return NULL; } rv = PyBytes_FromStringAndSize(NULL, (fragment->len/width)*newwidth); if (rv == NULL) return NULL; ncp = (unsigned char *)PyBytes_AsString(rv); for (i = j = 0; i < fragment->len; i += width, j += newwidth) { int val = GETSAMPLE32(width, fragment->buf, i); SETSAMPLE32(newwidth, ncp, j, val); } return rv; } static int gcd(int a, int b) { while (b > 0) { int tmp = a % b; a = b; b = tmp; } return a; } /*[clinic input] audioop.ratecv fragment: Py_buffer width: int nchannels: int inrate: int outrate: int state: object weightA: int = 1 weightB: int = 0 / Convert the frame rate of the input fragment. [clinic start generated code]*/ static PyObject * audioop_ratecv_impl(PyObject *module, Py_buffer *fragment, int width, int nchannels, int inrate, int outrate, PyObject *state, int weightA, int weightB) /*[clinic end generated code: output=624038e843243139 input=aff3acdc94476191]*/ { char *cp, *ncp; Py_ssize_t len; int chan, d, *prev_i, *cur_i, cur_o; PyObject *samps, *str, *rv = NULL, *channel; int bytes_per_frame; if (!audioop_check_size(width)) return NULL; if (nchannels < 1) { PyErr_SetString(AudioopError, "# of channels should be >= 1"); return NULL; } if (width > INT_MAX / nchannels) { /* This overflow test is rigorously correct because both multiplicands are >= 1. Use the argument names from the docs for the error msg. */ PyErr_SetString(PyExc_OverflowError, "width * nchannels too big for a C int"); return NULL; } bytes_per_frame = width * nchannels; if (weightA < 1 || weightB < 0) { PyErr_SetString(AudioopError, "weightA should be >= 1, weightB should be >= 0"); return NULL; } assert(fragment->len >= 0); if (fragment->len % bytes_per_frame != 0) { PyErr_SetString(AudioopError, "not a whole number of frames"); return NULL; } if (inrate <= 0 || outrate <= 0) { PyErr_SetString(AudioopError, "sampling rate not > 0"); return NULL; } /* divide inrate and outrate by their greatest common divisor */ d = gcd(inrate, outrate); inrate /= d; outrate /= d; /* divide weightA and weightB by their greatest common divisor */ d = gcd(weightA, weightB); weightA /= d; weightB /= d; if ((size_t)nchannels > SIZE_MAX/sizeof(int)) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); return NULL; } prev_i = (int *) PyMem_Malloc(nchannels * sizeof(int)); cur_i = (int *) PyMem_Malloc(nchannels * sizeof(int)); if (prev_i == NULL || cur_i == NULL) { (void) PyErr_NoMemory(); goto exit; } len = fragment->len / bytes_per_frame; /* # of frames */ if (state == Py_None) { d = -outrate; for (chan = 0; chan < nchannels; chan++) prev_i[chan] = cur_i[chan] = 0; } else { if (!PyTuple_Check(state)) { PyErr_SetString(PyExc_TypeError, "state must be a tuple or None"); goto exit; } if (!PyArg_ParseTuple(state, "iO!;ratecv(): illegal state argument", &d, &PyTuple_Type, &samps)) goto exit; if (PyTuple_Size(samps) != nchannels) { PyErr_SetString(AudioopError, "illegal state argument"); goto exit; } for (chan = 0; chan < nchannels; chan++) { channel = PyTuple_GetItem(samps, chan); if (!PyTuple_Check(channel)) { PyErr_SetString(PyExc_TypeError, "ratecv(): illegal state argument"); goto exit; } if (!PyArg_ParseTuple(channel, "ii;ratecv(): illegal state argument", &prev_i[chan], &cur_i[chan])) { goto exit; } } } /* str <- Space for the output buffer. */ if (len == 0) str = PyBytes_FromStringAndSize(NULL, 0); else { /* There are len input frames, so we need (mathematically) ceiling(len*outrate/inrate) output frames, and each frame requires bytes_per_frame bytes. Computing this without spurious overflow is the challenge; we can settle for a reasonable upper bound, though, in this case ceiling(len/inrate) * outrate. */ /* compute ceiling(len/inrate) without overflow */ Py_ssize_t q = 1 + (len - 1) / inrate; if (outrate > PY_SSIZE_T_MAX / q / bytes_per_frame) str = NULL; else str = PyBytes_FromStringAndSize(NULL, q * outrate * bytes_per_frame); } if (str == NULL) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); goto exit; } ncp = PyBytes_AsString(str); cp = fragment->buf; for (;;) { while (d < 0) { if (len == 0) { samps = PyTuple_New(nchannels); if (samps == NULL) goto exit; for (chan = 0; chan < nchannels; chan++) PyTuple_SetItem(samps, chan, Py_BuildValue("(ii)", prev_i[chan], cur_i[chan])); if (PyErr_Occurred()) goto exit; /* We have checked before that the length * of the string fits into int. */ len = (Py_ssize_t)(ncp - PyBytes_AsString(str)); rv = PyBytes_FromStringAndSize (PyBytes_AsString(str), len); Py_DECREF(str); str = rv; if (str == NULL) goto exit; rv = Py_BuildValue("(O(iO))", str, d, samps); Py_DECREF(samps); Py_DECREF(str); goto exit; /* return rv */ } for (chan = 0; chan < nchannels; chan++) { prev_i[chan] = cur_i[chan]; cur_i[chan] = GETSAMPLE32(width, cp, 0); cp += width; /* implements a simple digital filter */ cur_i[chan] = (int)( ((double)weightA * (double)cur_i[chan] + (double)weightB * (double)prev_i[chan]) / ((double)weightA + (double)weightB)); } len--; d += outrate; } while (d >= 0) { for (chan = 0; chan < nchannels; chan++) { cur_o = (int)(((double)prev_i[chan] * (double)d + (double)cur_i[chan] * (double)(outrate - d)) / (double)outrate); SETSAMPLE32(width, ncp, 0, cur_o); ncp += width; } d -= inrate; } } exit: PyMem_Free(prev_i); PyMem_Free(cur_i); return rv; } /*[clinic input] audioop.lin2ulaw fragment: Py_buffer width: int / Convert samples in the audio fragment to u-LAW encoding. [clinic start generated code]*/ static PyObject * audioop_lin2ulaw_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=14fb62b16fe8ea8e input=2450d1b870b6bac2]*/ { unsigned char *ncp; Py_ssize_t i; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; rv = PyBytes_FromStringAndSize(NULL, fragment->len/width); if (rv == NULL) return NULL; ncp = (unsigned char *)PyBytes_AsString(rv); for (i = 0; i < fragment->len; i += width) { int val = GETSAMPLE32(width, fragment->buf, i); *ncp++ = st_14linear2ulaw(val >> 18); } return rv; } /*[clinic input] audioop.ulaw2lin fragment: Py_buffer width: int / Convert sound fragments in u-LAW encoding to linearly encoded sound fragments. [clinic start generated code]*/ static PyObject * audioop_ulaw2lin_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=378356b047521ba2 input=45d53ddce5be7d06]*/ { unsigned char *cp; signed char *ncp; Py_ssize_t i; PyObject *rv; if (!audioop_check_size(width)) return NULL; if (fragment->len > PY_SSIZE_T_MAX/width) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); return NULL; } rv = PyBytes_FromStringAndSize(NULL, fragment->len*width); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); cp = fragment->buf; for (i = 0; i < fragment->len*width; i += width) { int val = st_ulaw2linear16(*cp++) << 16; SETSAMPLE32(width, ncp, i, val); } return rv; } /*[clinic input] audioop.lin2alaw fragment: Py_buffer width: int / Convert samples in the audio fragment to a-LAW encoding. [clinic start generated code]*/ static PyObject * audioop_lin2alaw_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=d076f130121a82f0 input=ffb1ef8bb39da945]*/ { unsigned char *ncp; Py_ssize_t i; PyObject *rv; if (!audioop_check_parameters(fragment->len, width)) return NULL; rv = PyBytes_FromStringAndSize(NULL, fragment->len/width); if (rv == NULL) return NULL; ncp = (unsigned char *)PyBytes_AsString(rv); for (i = 0; i < fragment->len; i += width) { int val = GETSAMPLE32(width, fragment->buf, i); *ncp++ = st_linear2alaw(val >> 19); } return rv; } /*[clinic input] audioop.alaw2lin fragment: Py_buffer width: int / Convert sound fragments in a-LAW encoding to linearly encoded sound fragments. [clinic start generated code]*/ static PyObject * audioop_alaw2lin_impl(PyObject *module, Py_buffer *fragment, int width) /*[clinic end generated code: output=85c365ec559df647 input=4140626046cd1772]*/ { unsigned char *cp; signed char *ncp; Py_ssize_t i; int val; PyObject *rv; if (!audioop_check_size(width)) return NULL; if (fragment->len > PY_SSIZE_T_MAX/width) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); return NULL; } rv = PyBytes_FromStringAndSize(NULL, fragment->len*width); if (rv == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(rv); cp = fragment->buf; for (i = 0; i < fragment->len*width; i += width) { val = st_alaw2linear16(*cp++) << 16; SETSAMPLE32(width, ncp, i, val); } return rv; } /*[clinic input] audioop.lin2adpcm fragment: Py_buffer width: int state: object / Convert samples to 4 bit Intel/DVI ADPCM encoding. [clinic start generated code]*/ static PyObject * audioop_lin2adpcm_impl(PyObject *module, Py_buffer *fragment, int width, PyObject *state) /*[clinic end generated code: output=cc19f159f16c6793 input=12919d549b90c90a]*/ { signed char *ncp; Py_ssize_t i; int step, valpred, delta, index, sign, vpdiff, diff; PyObject *rv = NULL, *str; int outputbuffer = 0, bufferstep; if (!audioop_check_parameters(fragment->len, width)) return NULL; /* Decode state, should have (value, step) */ if ( state == Py_None ) { /* First time, it seems. Set defaults */ valpred = 0; index = 0; } else if (!PyTuple_Check(state)) { PyErr_SetString(PyExc_TypeError, "state must be a tuple or None"); return NULL; } else if (!PyArg_ParseTuple(state, "ii;lin2adpcm(): illegal state argument", &valpred, &index)) { return NULL; } else if (valpred >= 0x8000 || valpred < -0x8000 || (size_t)index >= Py_ARRAY_LENGTH(stepsizeTable)) { PyErr_SetString(PyExc_ValueError, "bad state"); return NULL; } str = PyBytes_FromStringAndSize(NULL, fragment->len/(width*2)); if (str == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(str); step = stepsizeTable[index]; bufferstep = 1; for (i = 0; i < fragment->len; i += width) { int val = GETSAMPLE32(width, fragment->buf, i) >> 16; /* Step 1 - compute difference with previous value */ if (val < valpred) { diff = valpred - val; sign = 8; } else { diff = val - valpred; sign = 0; } /* Step 2 - Divide and clamp */ /* Note: ** This code *approximately* computes: ** delta = diff*4/step; ** vpdiff = (delta+0.5)*step/4; ** but in shift step bits are dropped. The net result of this ** is that even if you have fast mul/div hardware you cannot ** put it to good use since the fixup would be too expensive. */ delta = 0; vpdiff = (step >> 3); if ( diff >= step ) { delta = 4; diff -= step; vpdiff += step; } step >>= 1; if ( diff >= step ) { delta |= 2; diff -= step; vpdiff += step; } step >>= 1; if ( diff >= step ) { delta |= 1; vpdiff += step; } /* Step 3 - Update previous value */ if ( sign ) valpred -= vpdiff; else valpred += vpdiff; /* Step 4 - Clamp previous value to 16 bits */ if ( valpred > 32767 ) valpred = 32767; else if ( valpred < -32768 ) valpred = -32768; /* Step 5 - Assemble value, update index and step values */ delta |= sign; index += indexTable[delta]; if ( index < 0 ) index = 0; if ( index > 88 ) index = 88; step = stepsizeTable[index]; /* Step 6 - Output value */ if ( bufferstep ) { outputbuffer = (delta << 4) & 0xf0; } else { *ncp++ = (delta & 0x0f) | outputbuffer; } bufferstep = !bufferstep; } rv = Py_BuildValue("(O(ii))", str, valpred, index); Py_DECREF(str); return rv; } /*[clinic input] audioop.adpcm2lin fragment: Py_buffer width: int state: object / Decode an Intel/DVI ADPCM coded fragment to a linear fragment. [clinic start generated code]*/ static PyObject * audioop_adpcm2lin_impl(PyObject *module, Py_buffer *fragment, int width, PyObject *state) /*[clinic end generated code: output=3440ea105acb3456 input=f5221144f5ca9ef0]*/ { signed char *cp; signed char *ncp; Py_ssize_t i, outlen; int valpred, step, delta, index, sign, vpdiff; PyObject *rv, *str; int inputbuffer = 0, bufferstep; if (!audioop_check_size(width)) return NULL; /* Decode state, should have (value, step) */ if ( state == Py_None ) { /* First time, it seems. Set defaults */ valpred = 0; index = 0; } else if (!PyTuple_Check(state)) { PyErr_SetString(PyExc_TypeError, "state must be a tuple or None"); return NULL; } else if (!PyArg_ParseTuple(state, "ii;adpcm2lin(): illegal state argument", &valpred, &index)) { return NULL; } else if (valpred >= 0x8000 || valpred < -0x8000 || (size_t)index >= Py_ARRAY_LENGTH(stepsizeTable)) { PyErr_SetString(PyExc_ValueError, "bad state"); return NULL; } if (fragment->len > (PY_SSIZE_T_MAX/2)/width) { PyErr_SetString(PyExc_MemoryError, "not enough memory for output buffer"); return NULL; } outlen = fragment->len*width*2; str = PyBytes_FromStringAndSize(NULL, outlen); if (str == NULL) return NULL; ncp = (signed char *)PyBytes_AsString(str); cp = fragment->buf; step = stepsizeTable[index]; bufferstep = 0; for (i = 0; i < outlen; i += width) { /* Step 1 - get the delta value and compute next index */ if ( bufferstep ) { delta = inputbuffer & 0xf; } else { inputbuffer = *cp++; delta = (inputbuffer >> 4) & 0xf; } bufferstep = !bufferstep; /* Step 2 - Find new index value (for later) */ index += indexTable[delta]; if ( index < 0 ) index = 0; if ( index > 88 ) index = 88; /* Step 3 - Separate sign and magnitude */ sign = delta & 8; delta = delta & 7; /* Step 4 - Compute difference and new predicted value */ /* ** Computes 'vpdiff = (delta+0.5)*step/4', but see comment ** in adpcm_coder. */ vpdiff = step >> 3; if ( delta & 4 ) vpdiff += step; if ( delta & 2 ) vpdiff += step>>1; if ( delta & 1 ) vpdiff += step>>2; if ( sign ) valpred -= vpdiff; else valpred += vpdiff; /* Step 5 - clamp output value */ if ( valpred > 32767 ) valpred = 32767; else if ( valpred < -32768 ) valpred = -32768; /* Step 6 - Update step value */ step = stepsizeTable[index]; /* Step 6 - Output value */ SETSAMPLE32(width, ncp, i, valpred << 16); } rv = Py_BuildValue("(O(ii))", str, valpred, index); Py_DECREF(str); return rv; } #include "clinic/audioop.c.h" static PyMethodDef audioop_methods[] = { AUDIOOP_MAX_METHODDEF AUDIOOP_MINMAX_METHODDEF AUDIOOP_AVG_METHODDEF AUDIOOP_MAXPP_METHODDEF AUDIOOP_AVGPP_METHODDEF AUDIOOP_RMS_METHODDEF AUDIOOP_FINDFIT_METHODDEF AUDIOOP_FINDMAX_METHODDEF AUDIOOP_FINDFACTOR_METHODDEF AUDIOOP_CROSS_METHODDEF AUDIOOP_MUL_METHODDEF AUDIOOP_ADD_METHODDEF AUDIOOP_BIAS_METHODDEF AUDIOOP_ULAW2LIN_METHODDEF AUDIOOP_LIN2ULAW_METHODDEF AUDIOOP_ALAW2LIN_METHODDEF AUDIOOP_LIN2ALAW_METHODDEF AUDIOOP_LIN2LIN_METHODDEF AUDIOOP_ADPCM2LIN_METHODDEF AUDIOOP_LIN2ADPCM_METHODDEF AUDIOOP_TOMONO_METHODDEF AUDIOOP_TOSTEREO_METHODDEF AUDIOOP_GETSAMPLE_METHODDEF AUDIOOP_REVERSE_METHODDEF AUDIOOP_BYTESWAP_METHODDEF AUDIOOP_RATECV_METHODDEF { 0, 0 } }; static struct PyModuleDef audioopmodule = { PyModuleDef_HEAD_INIT, "audioop", NULL, -1, audioop_methods, NULL, NULL, NULL, NULL }; PyMODINIT_FUNC PyInit_audioop(void) { PyObject *m, *d; m = PyModule_Create(&audioopmodule); if (m == NULL) return NULL; d = PyModule_GetDict(m); if (d == NULL) return NULL; AudioopError = PyErr_NewException("audioop.error", NULL, NULL); if (AudioopError != NULL) PyDict_SetItemString(d,"error",AudioopError); return m; }